Re: [asterisk-dev] 2 way Audiohook ?

Okay, thanks, so I'll have to get the other channel and apply an audiohook to it Killian Matter Le jeu. 14 janv. 2021 à 14:42, Joshua C. Colp a écrit : > On Thu, Jan 14, 2021 at 9:40 AM Killian Matter > wrote: > >> Is it possible to have a 2 way Audiohook, that takes a

Re: [asterisk-dev] Help, can't get out of while loop

It seems that nothing else change it's status, i ran the loop for more than 15 minutes and still, no change of status. Killian Matter Le jeu. 14 janv. 2021 à 10:02, Dennis Buteyn a écrit : > That loop is equivalent to: > > audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN; > w

Re: [asterisk-dev] Help, can't get out of while loop

No, no media flowing Le jeu. 14 janv. 2021 à 11:15, Joshua C. Colp a écrit : > On Thu, Jan 14, 2021 at 6:10 AM Killian Matter > wrote: > >> It seems that nothing else change it's status, i ran the loop for more >> than 15 minutes and still, no change of status. >>

[asterisk-dev] 2 way Audiohook ?

take it. Thanks, Killian Matter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [asterisk-dev] Help, can't get out of while loop

Is it the same for datastore ? Le jeu. 14 janv. 2021 à 11:51, Joshua C. Colp a écrit : > On Thu, Jan 14, 2021 at 6:48 AM Killian Matter > wrote: > >> i'm trying to detach the audiohook after an h extension that's the >> origin of the problem, the flow of RTP just stop

[asterisk-dev] Help, can't get out of while loop

Hello , I'm developing a module on asterisk, while debugging i've come across a problem I don't quite understand. I'm using a noise filter, at the end of the call I stop my filter , so clean up everything, detach the audiohook and there is the problem. It's stuck in the while loop in

Re: [asterisk-dev] Help, can't get out of while loop

Okay, for now it's works ! Thanks for the help ! Killian Matter Le jeu. 14 janv. 2021 à 11:55, Joshua C. Colp a écrit : > On Thu, Jan 14, 2021 at 6:53 AM Killian Matter > wrote: > >> Is it the same for datastore ? >> > > Yes, datastores are also automatically des

Re: [asterisk-dev] Help, can't get out of while loop

by media flow you meant to allow media stream no ? I forgot to say i use SIP. Le jeu. 14 janv. 2021 à 11:33, Joshua C. Colp a écrit : > On Thu, Jan 14, 2021 at 6:31 AM Killian Matter > wrote: > >> No, no media flowing >> > > Audiohooks predate timers, and require

Re: [asterisk-dev] Help, can't get out of while loop

i'm trying to detach the audiohook after an h extension that's the origin of the problem, the flow of RTP just stop before. Have to find something else than the h extension, might try events. Le jeu. 14 janv. 2021 à 11:43, Joshua C. Colp a écrit : > On Thu, Jan 14, 2021 at 6:39 AM Kill

[asterisk-dev] Call and seg fault

Well hello, when the call is up, working, then asterisk put up a seg fault error. Just Segmentation Fault. What info would you want/do you need to help ? Killian MATTER -- _ -- Bandwidth and Colocation Provided by http

[asterisk-dev] Strange seg fault

look like a sync problem but is it even possible ? (e didn't ever see a sync prob in C language) Killian MATTER -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE

Re: [asterisk-dev] Strange seg fault

it doesn't have much time to detach audiohook and the dialplan goes on so some data are freed before the detach of audiohook and datastore do their job and so seg fault. Killian MATTER Le lun. 18 janv. 2021 à 12:08, Joshua C. Colp a écrit : > On Mon, Jan 18, 2021 at 7:00 AM Killian Matter >

[asterisk-dev] Segmentation fault on call

try to get access to something i shouldn't so i don't know at this point. Killian MATTER -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-dev] Call and seg fault

Sorry, I had School last, back to work this week, so i processed the core file. Looked into it, but there so much info, should something about the seg fault catch my eyes in those files ? Killian MATTER Le dim. 24 janv. 2021 à 09:27, Dennis Buteyn a écrit : > On 1/24/21 10:19 AM, LSV wr

[asterisk-dev] Struggling to get outbound channel

Hello ! forgive me, i'm struggling to get the outbound channel or the channels of a call. I already have the inbound channel, is it possible to go from the inbound channel and get either way the other channel or the whole lot of info about the two channels of the call ? Killian MATTER

Re: [asterisk-dev] Struggling to get outbound channel

Either way i try to catch the event when the bridge is done or like you said get the bridge and iterate the channels on it Killian MATTER Le mer. 3 févr. 2021 à 11:43, Joshua C. Colp a écrit : > On Wed, Feb 3, 2021 at 6:40 AM Killian Matter > wrote: > >> Hello ! >&g

Re: [asterisk-dev] "Echo call"

That's it, it should be fine with this thanks ! I feel dumb not finding it sooner Le ven. 26 mars 2021 à 12:11, Joshua C. Colp a écrit : > On Fri, Mar 26, 2021 at 8:09 AM Killian Matter > wrote: > >> Hello, I'm digging my head to find a way so that in a call as the >>

[asterisk-dev] "Echo call"

Hello, I'm digging my head to find a way so that in a call as the user/caller speak, he hear himself back, always having his echo of what he say. I thought I record the call then playback the record but I need the echo in live as he speaks. Thanks ! k.m --

[asterisk-dev] Calling an URL

Hello, I was wondering if it was possible to call an URL in VOIP with asterisk, is it? Thanks ! k.m -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

Re: [asterisk-dev] Use count

That's my plan, had hoped you could help, but thanks for your reactivity and your answers ! Best regards Le ven. 1 oct. 2021 à 15:59, Joshua C. Colp a écrit : > On Fri, Oct 1, 2021 at 10:54 AM Killian Matter > wrote: > >> It an app with 2 dialplan function, one to setup a fil

[asterisk-dev] Use count

Hello ! I wanted to ask, is there a way to put the use count of my custom module to 0 so that it's possible to unload without forcing ? Becaus once I've made a call, impossible to unload cause use count isn't 0. Thanks, K.m --

Re: [asterisk-dev] Use count

I don't manage it so it's through the dialplan ... Even so it's useless to try to decrease it through an API call ? Le ven. 1 oct. 2021 à 15:37, Joshua C. Colp a écrit : > On Fri, Oct 1, 2021 at 10:35 AM Killian Matter > wrote: > >> Hello ! >> I wanted to ask, is there

Re: [asterisk-dev] Use count

count isn't at 0 ? Le ven. 1 oct. 2021 à 15:48, Joshua C. Colp a écrit : > On Fri, Oct 1, 2021 at 10:42 AM Killian Matter > wrote: > >> I don't manage it so it's through the dialplan ... Even so it's useless >> to try to decrease it through an API call ? >> >